Multiplexing is done on IP flow basis which is characterized by source IPs and destination IP addresses. For each IP source/IP destination {IPS; IPD} flow between two IP hosts an own multiplexing stream is maintained.
Bandwidth saving is achieved by multiplexing several IP packets of an {IPS; IPD} flow into one IPM multiplexing packet removing the IP headers of the inserted IP packets. As an option, the RTP (Real-Time Protocol) header may be compressed in addition.
The more packets per sampling interval is can be multiplexed, the better is the bandwidth gain and the lower is transmission cost.
Current node implementations consist of multiple IP hosts (n). Consequently, m nodes with n IP hosts each would have n×m multiplexing streams and the probability for a sufficient number of IP packets/ts decreases. One option would be to increase is until sufficient IP packets are available per ts interval for multiplexing.
For real time services like voice, facsimile and circuit switched data in telecom networks end-to-end delay is a critical parameter which has to be kept to a minimum to sustain speech quality. End-to-end delay depends on delay generated due to coding and decoding, transmission delay in the IP backbone and multiplexing sampling time ts.
In order to sustain telecom grade speech quality is must be minimized. This means that ts and bandwidth demand compete and if both shall be minimized the number of IP hosts has to be minimal.
In the following an IP based telecom network example is described in connection with FIGS. 1 and 2, one time with a minimized multiplexing time interval and the other time the number of IP packets being maximized. The following example bases on a network with 10 million subscribers, the network having ten sites, a site corresponding e.g. to a town. The traffic per site indicating the number of calls taking place at the same time is supposed to be 20000 Erlang. Furthermore, it is supposed that 60% of the traffic stays within the site resulting in a traffic leaving the site through an IP multiplexer of 8000 Erlang. Furthermore, two media gateways per site are used in the example, meaning that non-site local traffic leaves the site through two media gateways. In FIG. 1 a table is shown indicating the gain for a first multiplexing time interval of 0.003 ms. For real-time applications this short multiplexing interval is advantageous. In the table shown in FIG. 1 the gain is indicated depending on the fact whether a RTP (Real-Time Transport Protocol) header compression is used or not. If all media gateways have 10 IP addresses, we get 3600 MUX (multiplexer) streams from each site. Each stream handled 72.22 packets per second, which means that during the multiplexing time interval is only one packet is collected. Thus, no multiplexing gain is achieved. It is even negative without RTP header compression. From FIG. 1 it can be concluded that a media gateway should contain one or maximal two IP hosts.
In FIG. 2 the same table is shown in which the multiplexing time interval is increased to 150 ms. In this case a bandwidth gain can already be obtained for 10 IP hosts per media gateway. However, the increased multiplexing time of 150 ms is normally not acceptable as delay for real-time connections. When more media gateways and more IP addresses per media gateway are used, the IP MUX stream number dramatically increases and the multiplexing gain decreases.
Accordingly, one problem can be seen in the fact that IP multiplexing is done per IP source/destination stream. In the case of many media gateways and many IP addresses per media gateway the number of IP packets which can be multiplexed in a specific time interval is very low and the proposed gain of 50% reduction of bandwidth is not reachable. On the other hand, it is not possible to increase the sampling rate for a bandwidth reduction, as the entire sampling rate is not acceptable for real-time applications, such as voice, facsimile or circuit switched data. This is applicable for mobile, wire line and radio networks.